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pjsip
client with psjip be called by other client,sip server receive "488 Not Acceptable Here" by other sip client.
Describe the bug
my sip client with psjip,can call other sip client,but when my sip client with psjip be called by other client,sip server receive "488 Not Acceptable Here" by my client with pjsip,is that any setting need I do?
Could anyone help me? thanks in advance!
Steps to reproduce
I try to use two client with pjsip call each other ,but alse one client will recieve the "488 Not Acceptable Here" . while the pjsip client call other sip client in mobile phone,the dialog can be establish,but I can hear the voice of pjsip client .Only the voice of phone can be played on the pjsip client.
PJSIP version
I don't know what version
Context
ESP32
Log, call stack, etc
**00:12:55.706 VoIPDemo ..Incoming call from <sip:[email protected]>!!
00:12:55.711 pjsua_call.c ..Answering call 2: code=200
00:12:55.718 inv0x3f837514 ....SDP negotiation done: Success
W (776837) pjsua_media.c: call_id = 2
00:12:55.726 pjsua_media.c .....Call 2: updating media..
W (776847) pjsua_media.c: for mi = 0
W (776847) pjsua_media.c: nn status = 0 call_med->type== 1
00:12:55.742 pjsua_media.c ......Call 2: stream #0 (audio) unchanged.
W (776857) pjsua_media.c: nn status = 0
00:12:55.753 pjsua_media.c ......Audio updated, stream #0: (inactive)
W (776867) pjsua_call.c: call->index = 2
00:12:55.764 pjsua_call.c .....Unable to create media session: Unknown pjmedia error 220048
[status=220048]
00:12:55.775 pjsua_core.c ........TX 487 bytes Response msg 488/INVITE/cseq=1652442112 (tdta0x3f83e0c8) to UDP 192.168.1.107:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH3200061c3e2318e0
Call-ID: 17e74ab7d786943606712597a901e1fbD20220513
From: <sip:[email protected]>;tag=f8070a9c72a10b09
To: <sip:[email protected]>;tag=sQM1w4T4o4KabwDR7hUV3SIo7w7v38Cu
CSeq: 1652442112 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0
--end msg--
00:12:55.831 VoIPDemo ...........Call 2 state=DISCONNCTD
00:12:55.835 pjsua_media.c ...........Call 2: deinitializing media..
00:12:55.842 pjsua_media.c .............Media stream call02:0 is destroyed
00:12:55.850 pjsua_call.c ...Error creating response: Unknown pjsip error 170488 [status=170488]
00:12:55.952 pjsua_core.c .RX 341 bytes Request msg ACK/cseq=1652442112 (rdata0x3f83ee88) from UDP 192.168.1.107:5060:
ACK sip:[email protected];ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH3200061c3e2318e0;rport
To: <sip:[email protected]>;tag=sQM1w4T4o4KabwDR7hUV3SIo7w7v38Cu
From: <sip:[email protected]>;tag=f8070a9c72a10b09
Call-ID: 17e74ab7d786943606712597a901e1fbD20220513
CSeq: 1652442112 ACK
Max-Forwards: 70
Content-Length: 0
--end msg--**
//the log that PC endpoint MicroSIP call the PJSIP g711 client ,the message packet:
INVITE sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXHf2611c929265d0bf;rport Contact: sip:[email protected] Call-ID: e9bf97655a136db816cee07179f7dbffD20220514 To: sip:[email protected] From: sip:[email protected];tag=34d8169a6e9c330c P-Asserted-Identity: sip:[email protected] Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Supported: 100rel User-Agent: miniSIPServer V39 (5 clients) build 20220417 CSeq: 1652511269 INVITE Content-Type: application/sdp Content-Length: 409
v=0 o=- 1652511269 1652511269 IN IP4 192.168.1.107 s=mss t=0 0 a=X-nat:0 m=audio 11006 RTP/AVP 8 0 101 c=IN IP4 192.168.1.107 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+Z3/0O95f9G4symYXUmZ5TJ44gZgRAXXQAXHZQFs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:MMhE4PUSf5q7qr6/Soc/D9yeHzcwJ/3ngjm6QvD1 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXHf2611c929265d0bf Call-ID: e9bf97655a136db816cee07179f7dbffD20220514 From: sip:[email protected];tag=34d8169a6e9c330c To: sip:[email protected] CSeq: 1652511269 INVITE Content-Length: 0
why I can not receive the message "SIP/2.0 180 Ringing" ? Is this normal?
I guess the pjsip sip client routine miss the project of "SIP/2.0 180 Ringing". But I have no idea how to correct it.
/**
* @hideinitializer
* No media is active after negotiation.
*/
#define PJMEDIA_SDPNEG_ENOMEDIA (PJMEDIA_ERRNO_START+48) /* 220048 */
Media codec not match between caller and callee.
Please confirm two sip clients use the same audio/video codec.
i Thanks for your reply. Let two the same device with PJSIP in G.711 call each other.They can not establish dialog successfully. They have the same codec.
i Thanks for your reply. Let two the same device with PJSIP in G.711 call each other.They can not establish dialog successfully. They have the same codec.
Same error? Could you supply full log?
i Thanks for your reply. Let two the same device with PJSIP in G.711 call each other.They can not establish dialog successfully. They have the same codec.
Same error? Could you supply full log?
@jimying Yes,it is the same error, why I can not receive the message "SIP/2.0 180 Ringing" ? Is this normal? that the message log I catch from sip server:
INVITE sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH8566a3de3ac0aa3f;rport Contact: sip:[email protected] Call-ID: 7cba50b7138cd764bc90c578dd326a2eD20220514 To: sip:[email protected] From: sip:[email protected];tag=6c4f84110dfb847e P-Asserted-Identity: sip:[email protected] Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Supported: 100rel User-Agent: miniSIPServer V39 (5 clients) build 20220417 CSeq: 1652515512 INVITE Content-Type: application/sdp Content-Length: 230
v=0 o=- 1652515512 1652515512 IN IP4 192.168.1.107 s=mss t=0 0 m=audio 11010 RTP/AVP 0 8 120 c=IN IP4 192.168.1.107 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH8566a3de3ac0aa3f Call-ID: 7cba50b7138cd764bc90c578dd326a2eD20220514 From: sip:[email protected];tag=6c4f84110dfb847e To: sip:[email protected] CSeq: 1652515512 INVITE Content-Length: 0
//if I do not answer the call the log is stop here
SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH8566a3de3ac0aa3f Call-ID: 7cba50b7138cd764bc90c578dd326a2eD20220514 From: sip:[email protected];tag=6c4f84110dfb847e To: sip:[email protected];tag=-YMsdAhygGRo9FFSJhTP-l5GWYaA6gSD CSeq: 1652515512 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Length: 0
ACK sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH8566a3de3ac0aa3f;rport To: sip:[email protected];tag=-YMsdAhygGRo9FFSJhTP-l5GWYaA6gSD From: sip:[email protected];tag=6c4f84110dfb847e Call-ID: 7cba50b7138cd764bc90c578dd326a2eD20220514 CSeq: 1652515512 ACK Max-Forwards: 70 Content-Length: 0
@CAIJUN111 You only supply sip message. I mean you should attach pjsip client's full log.
@CAIJUN111 You only supply sip message. I mean you supply pjsip client's full log.
@jimying oh,that the call's log(from register to invite) :
(1251) VoIPDemo: [1.0] Initialize peripherals management
I (1251) VoIPDemo: [1.1] Initialize and start peripherals
I (1251) VoIPDemo: [1.2] Start and wait for Wi-Fi network
I (1281) wifi:wifi driver task: 3ffcb218, prio:23, stack:6656, core=0
I (1281) system_api: Base MAC address is not set
I (1281) system_api: read default base MAC address from EFUSE
I (1291) wifi:wifi firmware version: 7679c42
I (1291) wifi:wifi certification version: v7.0
I (1291) wifi:config NVS flash: enabled
I (1291) wifi:config nano formating: disabled
I (1301) wifi:Init data frame dynamic rx buffer num: 128
I (1301) wifi:Init management frame dynamic rx buffer num: 128
I (1311) wifi:Init management short buffer num: 32
I (1311) wifi:Init static tx buffer num: 9
I (1321) wifi:Init tx cache buffer num: 32
I (1321) wifi:Init static rx buffer size: 1600
I (1321) wifi:Init static rx buffer num: 9
I (1331) wifi:Init dynamic rx buffer num: 128
I (1331) wifi_init: rx ba win: 16
I (1341) wifi_init: tcpip mbox: 64
I (1341) wifi_init: udp mbox: 64
I (1341) wifi_init: tcp mbox: 64
I (1351) wifi_init: tcp tx win: 5744
I (1351) wifi_init: tcp rx win: 5744
I (1361) wifi_init: tcp mss: 1436
I (1361) wifi_init: WiFi/LWIP prefer SPIRAM
I (1371) wifi_init: WiFi IRAM OP enabled
I (1371) wifi_init: WiFi RX IRAM OP enabled
I (1381) wifi_init: LWIP IRAM OP enabled
I (1381) phy_init: phy_version 4670,719f9f6,Feb 18 2021,17:07:07
I (1481) wifi:mode : sta (44:17:93:fa:81:e0)
I (1481) wifi:enable tsf
I (2211) wifi:new:<6,2>, old:<1,0>, ap:<255,255>, sta:<6,2>, prof:1
I (2781) wifi:state: init -> auth (b0)
I (2801) wifi:state: auth -> assoc (0)
I (2811) wifi:state: assoc -> run (10)
W (2841) wifi:
W (2931) PERIPH_WIFI: WiFi Event cb, Unhandle event_base:WIFI_EVENT, event_id:4 I (2941) wifi:AP's beacon interval = 102400 us, DTIM period = 1 I (3751) esp_netif_handlers: sta ip: 192.168.1.102, mask: 255.255.255.0, gw: 192.168.1.1 I (3751) PERIPH_WIFI: Got ip:192.168.1.102 I (3751) VoIPDemo: [ 2 ] Start codec chip W (4751) SPI: MCS ret:0,Status:15 I (4751) gpio: GPIO[22]| InputEn: 0| OutputEn: 1| OpenDrain: 0| Pullup: 0| Pulldown: 0| Intr:0 I (4751) gpio: GPIO[21]| InputEn: 0| OutputEn: 1| OpenDrain: 0| Pullup: 0| Pulldown: 0| Intr:0 W (4761) AUDIO_HAL: Codec mode is 3, Ctrl:1 I (4761) VoIPDemo: Func:app_main, Line:351, MEM Total:4299799 Bytes, Inter:210067 Bytes, Dram:208887 Bytes
I (4771) VoIPDemo: Func:app_main, Line:357, MEM Total:4299799 Bytes, Inter:210067 Bytes, Dram:208887 Bytes
app_main 0
sip_main 1
W (4791) pjsua_core.c: pjsua_create
W (4791) pjsua_core.c: init_data
00:00:06.267 os_core_esp32. !pjlib 1.0 for POSIX initialized
00:00:06.275 pjlib .select() I/O Queue created (0x3f82682c)
00:00:06.276 sip_endpoint.c .Module "mod-msg-print" registered
00:00:06.280 pjsua_core.c .PJSUA state changed: NULL --> CREATED
W (4821) pjsua_core.c: pjsua_reconfigure_logging
00:00:06.291 sip_endpoint.c .Module "mod-pjsua-log" registered
00:00:06.297 sip_endpoint.c .Module "mod-tsx-layer" registered
00:00:06.302 sip_endpoint.c .Module "mod-stateful-util" registered
00:00:06.308 sip_endpoint.c .Module "mod-ua" registered
00:00:06.313 sip_endpoint.c .Module "mod-100rel" registered
00:00:06.319 sip_endpoint.c .Module "mod-pjsua" registered
W (4861) pjsua_call.c: pjsua_call_subsys_init PJ_ARRAY_SIZE(pjsua_var->calls)12
00:00:06.336 sip_endpoint.c .Module "mod-invite" registered
00:00:06.337 esp_dev.c ..ESP driver found 1 devices
00:00:06.342 esp_dev.c ..ESP initialized
00:00:06.348 pjlib ..select() I/O Queue created (0x3f82c64c)
00:00:06.355 sip_endpoint.c .Module "mod-evsub" registered
00:00:06.359 sip_endpoint.c .Module "mod-mwi" registered
00:00:06.363 sip_endpoint.c .Module "mod-refer" registered
00:00:06.368 sip_endpoint.c .Module "mod-pjsua-options" registered
00:00:06.375 pjsua_core.c .1 SIP worker threads created
00:00:06.380 pjsua_core.c .pjsua version 1.0 initialized
00:00:06.385 pjsua_core.c .PJSUA state changed: CREATED --> INIT
W (4921) VoIPDemo: pjsua_transport_config_default:reuse_addr:1 cfg.public_addr.slen = 0
00:00:06.400 pjsua_core.c SIP UDP socket reachable at 192.168.1.102:5060
00:00:06.407 sip_transport_ Error setting SO_RCVBUF: Protocol not available
00:00:06.413 sip_transport_ Error setting SO_SNDBUF: Protocol not available
00:00:06.420 sip_transport. Remote address not registered, added the transport to the hash
00:00:06.428 sip_transport. Transport udp0x3f834004 registered: type=UDP, remote=:0
00:00:06.436 udp0x3f834004 SIP UDP transport started, published address is 192.168.1.102:5060W (4981) pjsua_core.c: Save the transport.
W (4981) pjsua_core.c: set_tp_state_cb
W (4981) pjsua_core.c: pjsua_start
00:00:06.457 pjsua_core.c PJSUA state changed: INIT --> STARTING
00:00:06.463 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
W (5001) pjsua_core.c: PJSUA_STATE_RUNNING
W (5011) pjsua_acc.c: pjsua_acc_add
00:00:06.478 pjsua_acc.c Adding account: id=sip:[email protected]
W (5021) pjsua_acc.c: initialize_acc
00:00:06.489 pjsua_acc.c .Account sip:[email protected] added with id 0
W (5031) pjsua_acc.c: pjsua_acc_set_registration
00:00:06.501 pjsua_acc.c .Acc 0: sting registration..
00:00:06.506 sip_transport. ..Acquiring transport type=UDP, sel=(null) remote=0.0.0.0:0
00:00:06.514 sip_transport. ..Transport udp0x3f834004 acquired
pjsua_acc_create_uac_contact status=0 0
pjsua_acc_create_uac_contact tp_type=1 1
pjsua_acc_create_uac_contact transport_param= 2
00:00:06.532 sip_transport. ..Acquiring transport type=UDP, sel=(null) remote=0.0.0.0:0
00:00:06.539 sip_transport. ..Transport udp0x3f834004 acquired
00:00:06.546 sip_transport. ...Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060
00:00:06.555 sip_transport. ...Transport udp0x3f834004 acquired
00:00:06.561 pjsua_core.c ...TX 482 bytes Request msg REGISTER/cseq=22357 (tdta0x3f837514) to UDP 192.168.1.107:5060:
REGISTER sip:192.168.1.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjatoJ4LRZt4HF2mZe2k.JgQBrsoiqctC9
Max-Forwards: 70
From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J
To: sip:[email protected]
Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k
CSeq: 22357 REGISTER
Contact: sip:[email protected]:5060;ob
Expires: 18000
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS
Content-Length: 0
--end msg--
00:00:06.618 pjsua_acc.c ..Acc 0: Registration sent
W (5171) wifi:
--end msg--
00:00:06.778 sip_transport. ....Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060
00:00:06.785 sip_transport. ....Transport udp0x3f834004 acquired
00:00:06.791 pjsua_core.c ....TX 675 bytes Request msg REGISTER/cseq=22358 (tdta0x3f837514)
to UDP 192.168.1.107:5060:
REGISTER sip:192.168.1.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjRvDhCbT06wVAPT374IttvJZlx.AihHPd
Max-Forwards: 70
From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J
To: sip:[email protected]
Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k
CSeq: 22358 REGISTER
Contact: sip:[email protected]:5060;ob
Expires: 18000
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS
Authorization: Digest username="111", realm="myvoipapp.com", nonce="57BB0AE23A0AE4234B9D1B3B4BAFD85A", uri="sip:192.168.1.107:5060", response="1cd80954ba86bad7a23da8450008813d", algorithm=MD5
Content-Length: 0
--end msg--
00:00:06.974 pjsua_core.c .RX 539 bytes Response msg 200/REGISTER/cseq=22358 (rdata0x3f83a278) from UDP 192.168.1.107:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjRvDhCbT06wVAPT374IttvJZlx.AihHPd;received=192.168.1.102;rport=5060
From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J
To: sip:[email protected];tag=81d00c8cb32ede8e
CSeq: 22358 REGISTER
Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: "111"sip:[email protected];ob
Server: miniSIPServer V39 (5 clients) build 20220417
Expires: 120
Content-Length: 0
--end msg--
00:00:07.025 pjsua_acc.c ....SIP outbound status for acc 0 is not active
00:00:07.031 pjsua_acc.c ....sip:[email protected]: registration success, status=200 (OK),
will re-register in 120 seconds
00:00:07.041 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:192.168.1.107:5060, interval:15s
W (5581) pjsua_acc.c: 1pjsua_acc_get_info acc_id = 0
W (5591) pjsua_acc.c: 2pjsua_acc_get_info
W (5591) THIS_FILE: 3pjsua_acc_get_info
W (5601) THIS_FILE: 4pjsua_acc_get_info
W (5601) THIS_FILE: 5pjsua_acc_get_info
W (5611) 5.5pjsua_acc_get_info: acc->reg_last_code = 200
W (5611) init_status_phrase: sizeof(status_phrase) = 5680 8
W (5621) 7pjsua_acc_get_info: info->status_text = OK
W (5621) THIS_FILE: 6pjsua_acc_get_info
W (5631) pjsua_acc.c: 3pjsua_acc_get_info
W (5631) VoIPDemo: on_reg_state2
on_reg_state2 account:sip:[email protected] logined ok.
00:00:07.111 pjsua_call.c ....Making call with acc #0 to sip:[email protected]:5060
00:00:07.120 pjsua_media.c .....Call 0: initializing media..
00:00:07.125 pjsua_media.c ......RTP socket reachable at 192.168.1.102:4000
00:00:07.131 pjsua_media.c ......RTCP socket reachable at 192.168.1.102:4001
00:00:07.139 pjsua_media.c ......Media index 0 selected for audio call 0
W (5681) pjsua_core.c: pjsua_process_msg_data
00:00:07.152 sip_transport. ........Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060
00:00:07.160 sip_transport. ........Transport udp0x3f834004 acquired
00:00:07.167 pjsua_core.c ........TX 910 bytes Request msg INVITE/cseq=21755 (tdta0x3f83f25c) to UDP 192.168.1.107:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs
Max-Forwards: 70
From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F
To: sip:[email protected]
Contact: sip:[email protected]:5060;ob
Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q
CSeq: 21755 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 331
v=0 o=- 2208988807 2208988807 IN IP4 192.168.1.102 s=pjmedia b=AS:84 t=0 0 m=audio 4000 RTP/AVP 0 8 120 c=IN IP4 192.168.1.102 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.1.102 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=ssrc:730417256 cname:2e533cc404185faf
--end msg--
00:00:07.262 VoIPDemo ...........Call 0 state=CALLING
00:00:07.374 pjsua_core.c .RX 406 bytes Response msg 100/INVITE/cseq=21755 (rdata0x3f841c34) from UDP 192.168.1.107:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs;received=192.168.1.102;rport=5060
From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F
To: sip:[email protected]
CSeq: 21755 INVITE
Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Content-Length: 0
--end msg-- 00:00:07.854 pjsua_core.c .RX 531 bytes Response msg 480/INVITE/cseq=21755 (rdata0x3f841c34) from UDP 192.168.1.107:5060: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs;received=192.168.1.102;rport=5060 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected];tag=2b8ebc674a425ebc CSeq: 21755 INVITE Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Contact: sip:[email protected] Reason: Q.850; cause=31; text="Normal, unspecified" Content-Length: 0
--end msg-- 00:00:07.905 pjsua_core.c ..TX 346 bytes Request msg ACK/cseq=21755 (tdta0x3f83a2c8) to UDP 192.168.1.107:5060: ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs Max-Forwards: 70 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected];tag=2b8ebc674a425ebc Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q CSeq: 21755 ACK Content-Length: 0
--end msg-- 00:00:07.947 VoIPDemo .....Call 0 state=DISCONNCTD 00:00:07.950 pjsua_media.c .....Call 0: deinitializing media.. 00:00:07.955 pjsua_media.c ......Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0
that the callee's log(just answer the call,they have the same register process): 00:16:58.882 sip_transport. Acquiring transport type=UDP, sel=transport[udp 0.0.0.0:5060 [published as 192.168.1.104:5060]], reuse=1 remote=192.168.1.107:5060 00:16:58.885 sip_transport. Transport udp0x3f834004 acquired 00:17:01.622 pjsua_core.c .RX 819 bytes Request msg INVITE/cseq=1652516407 (rdata0x3f8412bc) from UDP 192.168.1.107:5060: INVITE sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea;rport Contact: sip:[email protected] Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 To: sip:[email protected] From: sip:[email protected];tag=ca9bc129c89b3b4d P-Asserted-Identity: sip:[email protected] Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Supported: 100rel User-Agent: miniSIPServer V39 (5 clients) build 20220417 CSeq: 1652516407 INVITE Content-Type: application/sdp Content-Length: 230
v=0 o=- 1652516407 1652516407 IN IP4 192.168.1.107 s=mss t=0 0 m=audio 11006 RTP/AVP 0 8 120 c=IN IP4 192.168.1.107 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16
--end msg--
W (1022807) pjsua_call.c: pjsua_call_on_incoming
00:17:01.702 pjsua_call.c .Incoming Request msg INVITE/cseq=1652516407 (rdata0x3f8412bc)
00:17:01.718 pjsua_media.c ..Call 1: initializing media..
00:17:01.720 pjsua_media.c ...RTP socket reachable at 192.168.1.104:4018
00:17:01.722 pjsua_media.c ...RTCP socket reachable at 192.168.1.104:4019
00:17:01.730 pjsua_media.c ...Media index 0 selected for audio call 1
00:17:01.740 pjsua_core.c .....TX 300 bytes Response msg 100/INVITE/cseq=1652516407 (tdta0x3f843898) to UDP 192.168.1.107:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea
Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514
From: sip:[email protected];tag=ca9bc129c89b3b4d
To: sip:[email protected]
CSeq: 1652516407 INVITE
Content-Length: 0
--end msg--
00:17:01.777 VoIPDemo ..Incoming call from sip:[email protected]!!
00:7:01.781 pjsua_call.c ..Answering call 1: code=200
00:17:01.788 inv0x3f838c88 ....SDP negotiation done: Success
W (1022907) pjsua_media.c: call_id = 1
00:17:01.796 pjsua_media.c .....Call 1: updating media..
W (1022917) pjsua_media.c: for mi = 0
W (1022917) pjsua_media.c: nn status = 0 call_med->type== 1
00:17:01.812 pjsua_media.c ......Call 1: stream #0 (audio) unchanged.
W (1022927) pjsua_media.c: nn status = 0
00:17:01.823 pjsua_media.c ......Audio updated, stream #0: (inactive)
W (1022937) pjsua_call.c: call->index = 1
00:17:01.834 pjsua_call.c .....Unable to create media session: Unknown pjmedia error 220048
[status=220048]
00:17:01.845 pjsua_core.c ........TX 487 bytes Response msg 488/INVITE/cseq=1652516407 (tdta0x3f83ca08) to UDP 192.168.1.107:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea
Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514
From: sip:[email protected];tag=ca9bc129c89b3b4d
To: sip:[email protected];tag=kfCQasyuubZXzirZ6dsffJhxxzcb.oi5
CSeq: 1652516407 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0
--end msg-- 00:17:01.902 VoIPDemo ...........Call 1 state=DISCONNCTD 00:17:01.906 pjsua_media.c ...........Call 1: deinitializing media.. 00:17:01.912 pjsua_media.c .............Media stream call01:0 is destroyed 00:17:01.920 pjsua_call.c ...Error creating response: Unknown pjsip error 170488 [status=170488] 00:17:02.023 pjsua_core.c .RX 341 bytes Request msg ACK/cseq=1652516407 (rdata0x3f83d7c8) from UDP 192.168.1.107:5060: ACK sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea;rport To: sip:[email protected];tag=kfCQasyuubZXzirZ6dsffJhxxzcb.oi5 From: sip:[email protected];tag=ca9bc129c89b3b4d Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 CSeq: 1652516407 ACK Max-Forwards: 70 Content-Length: 0
--end msg--
@CAIJUN111 You only supply sip message. I mean you supply pjsip client's full log.
@jimying oh,that the call's log(from register to invite) :
(1251) VoIPDemo: [1.0] Initialize peripherals management I (1251) VoIPDemo: [1.1] Initialize and start peripherals I (1251) VoIPDemo: [1.2] Start and wait for Wi-Fi network I (1281) wifi:wifi driver task: 3ffcb218, prio:23, stack:6656, core=0 I (1281) system_api: Base MAC address is not set I (1281) system_api: read default base MAC address from EFUSE I (1291) wifi:wifi firmware version: 7679c42 I (1291) wifi:wifi certification version: v7.0 I (1291) wifi:config NVS flash: enabled I (1291) wifi:config nano formating: disabled I (1301) wifi:Init data frame dynamic rx buffer num: 128 I (1301) wifi:Init management frame dynamic rx buffer num: 128 I (1311) wifi:Init management short buffer num: 32 I (1311) wifi:Init static tx buffer num: 9 I (1321) wifi:Init tx cache buffer num: 32 I (1321) wifi:Init static rx buffer size: 1600 I (1321) wifi:Init static rx buffer num: 9 I (1331) wifi:Init dynamic rx buffer num: 128 I (1331) wifi_init: rx ba win: 16 I (1341) wifi_init: tcpip mbox: 64 I (1341) wifi_init: udp mbox: 64 I (1341) wifi_init: tcp mbox: 64 I (1351) wifi_init: tcp tx win: 5744 I (1351) wifi_init: tcp rx win: 5744 I (1361) wifi_init: tcp mss: 1436 I (1361) wifi_init: WiFi/LWIP prefer SPIRAM I (1371) wifi_init: WiFi IRAM OP enabled I (1371) wifi_init: WiFi RX IRAM OP enabled I (1381) wifi_init: LWIP IRAM OP enabled I (1381) phy_init: phy_version 4670,719f9f6,Feb 18 2021,17:07:07 I (1481) wifi:mode : sta (44:17:93:fa:81:e0) I (1481) wifi:enable tsf I (2211) wifi🆕<6,2>, old:<1,0>, ap:<255,255>, sta:<6,2>, prof:1 I (2781) wifi:state: init -> auth (b0) I (2801) wifi:state: auth -> assoc (0) I (2811) wifi:state: assoc -> run (10) W (2841) wifi:idx:0 (ifx:0, cc:08:fb:a8:93:55), tid:5, ssn:0, winSize:64 I (2921) wifi:connected with APLD-417-2.4G, aid = 7, channel 6, 40D, bssid = cc:08:fb:a8:93:55 I (2921) wifi:security: WPA2-PSK, phy: bgn, rssi: -50 I (2931) wifi:pm start, type: 1
W (2931) PERIPH_WIFI: WiFi Event cb, Unhandle event_base:WIFI_EVENT, event_id:4 I (2941) wifi:AP's beacon interval = 102400 us, DTIM period = 1 I (3751) esp_netif_handlers: sta ip: 192.168.1.102, mask: 255.255.255.0, gw: 192.168.1.1 I (3751) PERIPH_WIFI: Got ip:192.168.1.102 I (3751) VoIPDemo: [ 2 ] Start codec chip W (4751) SPI: MCS ret:0,Status:15 I (4751) gpio: GPIO[22]| InputEn: 0| OutputEn: 1| OpenDrain: 0| Pullup: 0| Pulldown: 0| Intr:0 I (4751) gpio: GPIO[21]| InputEn: 0| OutputEn: 1| OpenDrain: 0| Pullup: 0| Pulldown: 0| Intr:0 W (4761) AUDIO_HAL: Codec mode is 3, Ctrl:1 I (4761) VoIPDemo: Func:app_main, Line:351, MEM Total:4299799 Bytes, Inter:210067 Bytes, Dram:208887 Bytes
I (4771) VoIPDemo: Func:app_main, Line:357, MEM Total:4299799 Bytes, Inter:210067 Bytes, Dram:208887 Bytes
app_main 0 sip_main 1 W (4791) pjsua_core.c: pjsua_create W (4791) pjsua_core.c: init_data 00:00:06.267 os_core_esp32. !pjlib 1.0 for POSIX initialized 00:00:06.275 pjlib .select() I/O Queue created (0x3f82682c) 00:00:06.276 sip_endpoint.c .Module "mod-msg-print" registered 00:00:06.280 pjsua_core.c .PJSUA state changed: NULL --> CREATED W (4821) pjsua_core.c: pjsua_reconfigure_logging 00:00:06.291 sip_endpoint.c .Module "mod-pjsua-log" registered 00:00:06.297 sip_endpoint.c .Module "mod-tsx-layer" registered 00:00:06.302 sip_endpoint.c .Module "mod-stateful-util" registered 00:00:06.308 sip_endpoint.c .Module "mod-ua" registered 00:00:06.313 sip_endpoint.c .Module "mod-100rel" registered 00:00:06.319 sip_endpoint.c .Module "mod-pjsua" registered W (4861) pjsua_call.c: pjsua_call_subsys_init PJ_ARRAY_SIZE(pjsua_var->calls)12 00:00:06.336 sip_endpoint.c .Module "mod-invite" registered 00:00:06.337 esp_dev.c ..ESP driver found 1 devices 00:00:06.342 esp_dev.c ..ESP initialized 00:00:06.348 pjlib ..select() I/O Queue created (0x3f82c64c) 00:00:06.355 sip_endpoint.c .Module "mod-evsub" registered 00:00:06.359 sip_endpoint.c .Module "mod-mwi" registered 00:00:06.363 sip_endpoint.c .Module "mod-refer" registered 00:00:06.368 sip_endpoint.c .Module "mod-pjsua-options" registered 00:00:06.375 pjsua_core.c .1 SIP worker threads created 00:00:06.380 pjsua_core.c .pjsua version 1.0 initialized 00:00:06.385 pjsua_core.c .PJSUA state changed: CREATED --> INIT W (4921) VoIPDemo: pjsua_transport_config_default:reuse_addr:1 cfg.public_addr.slen = 0 00:00:06.400 pjsua_core.c SIP UDP socket reachable at 192.168.1.102:5060 00:00:06.407 sip_transport_ Error setting SO_RCVBUF: Protocol not available 00:00:06.413 sip_transport_ Error setting SO_SNDBUF: Protocol not available 00:00:06.420 sip_transport. Remote address not registered, added the transport to the hash 00:00:06.428 sip_transport. Transport udp0x3f834004 registered: type=UDP, remote=:0 00:00:06.436 udp0x3f834004 SIP UDP transport started, published address is 192.168.1.102:5060W (4981) pjsua_core.c: Save the transport. W (4981) pjsua_core.c: set_tp_state_cb W (4981) pjsua_core.c: pjsua_start 00:00:06.457 pjsua_core.c PJSUA state changed: INIT --> STARTING 00:00:06.463 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING W (5001) pjsua_core.c: PJSUA_STATE_RUNNING W (5011) pjsua_acc.c: pjsua_acc_add 00:00:06.478 pjsua_acc.c Adding account: id=sip:[email protected] W (5021) pjsua_acc.c: initialize_acc 00:00:06.489 pjsua_acc.c .Account sip:[email protected] added with id 0 W (5031) pjsua_acc.c: pjsua_acc_set_registration 00:00:06.501 pjsua_acc.c .Acc 0: sting registration.. 00:00:06.506 sip_transport. ..Acquiring transport type=UDP, sel=(null) remote=0.0.0.0:0 00:00:06.514 sip_transport. ..Transport udp0x3f834004 acquired pjsua_acc_create_uac_contact status=0 0 pjsua_acc_create_uac_contact tp_type=1 1 pjsua_acc_create_uac_contact transport_param= 2 00:00:06.532 sip_transport. ..Acquiring transport type=UDP, sel=(null) remote=0.0.0.0:0 00:00:06.539 sip_transport. ..Transport udp0x3f834004 acquired 00:00:06.546 sip_transport. ...Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060 00:00:06.555 sip_transport. ...Transport udp0x3f834004 acquired 00:00:06.561 pjsua_core.c ...TX 482 bytes Request msg REGISTER/cseq=22357 (tdta0x3f837514) to UDP 192.168.1.107:5060: REGISTER sip:192.168.1.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjatoJ4LRZt4HF2mZe2k.JgQBrsoiqctC9 Max-Forwards: 70 From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J To: sip:[email protected] Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k CSeq: 22357 REGISTER Contact: sip:[email protected]:5060;ob Expires: 18000 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS Content-Length: 0
--end msg-- 00:00:06.618 pjsua_acc.c ..Acc 0: Registration sent W (5171) wifi:idx:1 (ifx:0, cc:08:fb:a8:93:55), tid:0, ssn:1, winSize:64 00:00:06.724 pjsua_core.c !.RX 556 bytes Response msg 401/REGISTER/cseq=22357 (rdata0x3f834e28) from UDP 192.168.1.107:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjatoJ4LRZt4HF2mZe2k.JgQBrsoiqctC9;received=192.168.1.102;rport=5060 From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J To: sip:[email protected];tag=b4d22ade4d61b393 CSeq: 22357 REGISTER Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE WWW-Authenticate: Digest realm="myvoipapp.com", nonce="57BB0AE23A0AE4234B9D1B3B4BAFD85A", algorithm=MD5, stale=true Content-Length: 0
--end msg-- 00:00:06.778 sip_transport. ....Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060 00:00:06.785 sip_transport. ....Transport udp0x3f834004 acquired 00:00:06.791 pjsua_core.c ....TX 675 bytes Request msg REGISTER/cseq=22358 (tdta0x3f837514) to UDP 192.168.1.107:5060: REGISTER sip:192.168.1.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjRvDhCbT06wVAPT374IttvJZlx.AihHPd Max-Forwards: 70 From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J To: sip:[email protected] Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k CSeq: 22358 REGISTER Contact: sip:[email protected]:5060;ob Expires: 18000 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS Authorization: Digest username="111", realm="myvoipapp.com", nonce="57BB0AE23A0AE4234B9D1B3B4BAFD85A", uri="sip:192.168.1.107:5060", response="1cd80954ba86bad7a23da8450008813d", algorithm=MD5 Content-Length: 0
--end msg-- 00:00:06.974 pjsua_core.c .RX 539 bytes Response msg 200/REGISTER/cseq=22358 (rdata0x3f83a278) from UDP 192.168.1.107:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjRvDhCbT06wVAPT374IttvJZlx.AihHPd;received=192.168.1.102;rport=5060 From: sip:[email protected];tag=5FavHRDnpFKpfQUFlKrCval-4NNL463J To: sip:[email protected];tag=81d00c8cb32ede8e CSeq: 22358 REGISTER Call-ID: 5iiaRLvyuah4OBhaWIGcrjZTJtUHf24k Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Contact: "111"sip:[email protected];ob Server: miniSIPServer V39 (5 clients) build 20220417 Expires: 120 Content-Length: 0
--end msg-- 00:00:07.025 pjsua_acc.c ....SIP outbound status for acc 0 is not active 00:00:07.031 pjsua_acc.c ....sip:[email protected]: registration success, status=200 (OK), will re-register in 120 seconds 00:00:07.041 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:192.168.1.107:5060, interval:15s W (5581) pjsua_acc.c: 1pjsua_acc_get_info acc_id = 0 W (5591) pjsua_acc.c: 2pjsua_acc_get_info W (5591) THIS_FILE: 3pjsua_acc_get_info W (5601) THIS_FILE: 4pjsua_acc_get_info W (5601) THIS_FILE: 5pjsua_acc_get_info W (5611) 5.5pjsua_acc_get_info: acc->reg_last_code = 200 W (5611) init_status_phrase: sizeof(status_phrase) = 5680 8 W (5621) 7pjsua_acc_get_info: info->status_text = OK W (5621) THIS_FILE: 6pjsua_acc_get_info W (5631) pjsua_acc.c: 3pjsua_acc_get_info W (5631) VoIPDemo: on_reg_state2 on_reg_state2 account:sip:[email protected] logined ok. 00:00:07.111 pjsua_call.c ....Making call with acc #0 to sip:[email protected]:5060 00:00:07.120 pjsua_media.c .....Call 0: initializing media.. 00:00:07.125 pjsua_media.c ......RTP socket reachable at 192.168.1.102:4000 00:00:07.131 pjsua_media.c ......RTCP socket reachable at 192.168.1.102:4001 00:00:07.139 pjsua_media.c ......Media index 0 selected for audio call 0 W (5681) pjsua_core.c: pjsua_process_msg_data 00:00:07.152 sip_transport. ........Acquiring transport type=UDP, sel=unknown[0x0], reuse=1 remote=192.168.1.107:5060 00:00:07.160 sip_transport. ........Transport udp0x3f834004 acquired 00:00:07.167 pjsua_core.c ........TX 910 bytes Request msg INVITE/cseq=21755 (tdta0x3f83f25c) to UDP 192.168.1.107:5060: INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs Max-Forwards: 70 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected] Contact: sip:[email protected]:5060;ob Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q CSeq: 21755 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 331
v=0 o=- 2208988807 2208988807 IN IP4 192.168.1.102 s=pjmedia b=AS:84 t=0 0 m=audio 4000 RTP/AVP 0 8 120 c=IN IP4 192.168.1.102 b=TIAS:64000 a=rtcp:4001 IN IP4 192.168.1.102 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=ssrc:730417256 cname:2e533cc404185faf
--end msg-- 00:00:07.262 VoIPDemo ...........Call 0 state=CALLING 00:00:07.374 pjsua_core.c .RX 406 bytes Response msg 100/INVITE/cseq=21755 (rdata0x3f841c34) from UDP 192.168.1.107:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs;received=192.168.1.102;rport=5060 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected] CSeq: 21755 INVITE Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Content-Length: 0
--end msg-- 00:00:07.854 pjsua_core.c .RX 531 bytes Response msg 480/INVITE/cseq=21755 (rdata0x3f841c34) from UDP 192.168.1.107:5060: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs;received=192.168.1.102;rport=5060 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected];tag=2b8ebc674a425ebc CSeq: 21755 INVITE Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Contact: sip:[email protected] Reason: Q.850; cause=31; text="Normal, unspecified" Content-Length: 0
--end msg-- 00:00:07.905 pjsua_core.c ..TX 346 bytes Request msg ACK/cseq=21755 (tdta0x3f83a2c8) to UDP 192.168.1.107:5060: ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.102:5060;rport;branch=z9hG4bKPjn6eO4oHxKu28H5VRBZFZ3RKTYGA5-YMs Max-Forwards: 70 From: sip:[email protected];tag=FTMRp6Qbn.3tPd1XGgeWISiBkm3wld7F To: sip:[email protected];tag=2b8ebc674a425ebc Call-ID: x0KxpFxEc0Nwo7ZXf7gOEBHbNGYngf4Q CSeq: 21755 ACK Content-Length: 0
--end msg-- 00:00:07.947 VoIPDemo .....Call 0 state=DISCONNCTD 00:00:07.950 pjsua_media.c .....Call 0: deinitializing media.. 00:00:07.955 pjsua_media.c ......Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0
that the callee's log(just answer the call,they have the same register process): 00:16:58.882 sip_transport. Acquiring transport type=UDP, sel=transport[udp 0.0.0.0:5060 [published as 192.168.1.104:5060]], reuse=1 remote=192.168.1.107:5060 00:16:58.885 sip_transport. Transport udp0x3f834004 acquired 00:17:01.622 pjsua_core.c .RX 819 bytes Request msg INVITE/cseq=1652516407 (rdata0x3f8412bc) from UDP 192.168.1.107:5060: INVITE sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea;rport Contact: sip:[email protected] Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 To: sip:[email protected] From: sip:[email protected];tag=ca9bc129c89b3b4d P-Asserted-Identity: sip:[email protected] Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE Supported: 100rel User-Agent: miniSIPServer V39 (5 clients) build 20220417 CSeq: 1652516407 INVITE Content-Type: application/sdp Content-Length: 230
v=0 o=- 1652516407 1652516407 IN IP4 192.168.1.107 s=mss t=0 0 m=audio 11006 RTP/AVP 0 8 120 c=IN IP4 192.168.1.107 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16
--end msg-- W (1022807) pjsua_call.c: pjsua_call_on_incoming 00:17:01.702 pjsua_call.c .Incoming Request msg INVITE/cseq=1652516407 (rdata0x3f8412bc) 00:17:01.718 pjsua_media.c ..Call 1: initializing media.. 00:17:01.720 pjsua_media.c ...RTP socket reachable at 192.168.1.104:4018 00:17:01.722 pjsua_media.c ...RTCP socket reachable at 192.168.1.104:4019 00:17:01.730 pjsua_media.c ...Media index 0 selected for audio call 1 00:17:01.740 pjsua_core.c .....TX 300 bytes Response msg 100/INVITE/cseq=1652516407 (tdta0x3f843898) to UDP 192.168.1.107:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 From: sip:[email protected];tag=ca9bc129c89b3b4d To: sip:[email protected] CSeq: 1652516407 INVITE Content-Length: 0
--end msg-- 00:17:01.777 VoIPDemo ..Incoming call from sip:[email protected]!! 00:7:01.781 pjsua_call.c ..Answering call 1: code=200 00:17:01.788 inv0x3f838c88 ....SDP negotiation done: Success W (1022907) pjsua_media.c: call_id = 1 00:17:01.796 pjsua_media.c .....Call 1: updating media.. W (1022917) pjsua_media.c: for mi = 0 W (1022917) pjsua_media.c: nn status = 0 call_med->type== 1 00:17:01.812 pjsua_media.c ......Call 1: stream #0 (audio) unchanged. W (1022927) pjsua_media.c: nn status = 0 00:17:01.823 pjsua_media.c ......Audio updated, stream #0: (inactive) W (1022937) pjsua_call.c: call->index = 1 00:17:01.834 pjsua_call.c .....Unable to create media session: Unknown pjmedia error 220048 [status=220048] 00:17:01.845 pjsua_core.c ........TX 487 bytes Response msg 488/INVITE/cseq=1652516407 (tdta0x3f83ca08) to UDP 192.168.1.107:5060: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.1.107;rport=5060;received=192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 From: sip:[email protected];tag=ca9bc129c89b3b4d To: sip:[email protected];tag=kfCQasyuubZXzirZ6dsffJhxxzcb.oi5 CSeq: 1652516407 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Length: 0
--end msg-- 00:17:01.902 VoIPDemo ...........Call 1 state=DISCONNCTD 00:17:01.906 pjsua_media.c ...........Call 1: deinitializing media.. 00:17:01.912 pjsua_media.c .............Media stream call01:0 is destroyed 00:17:01.920 pjsua_call.c ...Error creating response: Unknown pjsip error 170488 [status=170488] 00:17:02.023 pjsua_core.c .RX 341 bytes Request msg ACK/cseq=1652516407 (rdata0x3f83d7c8) from UDP 192.168.1.107:5060: ACK sip:[email protected];ob SIP/2.0 Via: SIP/2.0/UDP 192.168.1.107;branch=z9hG4bKYXH1ae8b701701eccea;rport To: sip:[email protected];tag=kfCQasyuubZXzirZ6dsffJhxxzcb.oi5 From: sip:[email protected];tag=ca9bc129c89b3b4d Call-ID: 1fb563325676bc87628ba9a2327b9272D20220514 CSeq: 1652516407 ACK Max-Forwards: 70 Content-Length: 0
--end msg--
@jimying The receiver of this call, it never sent message "SIP/2.0 180 Ringing",is this reason of the errror?
NO
Callee log part here: 00:17:01.777 VoIPDemo ..Incoming call from sip:[email protected]!! 00:7:01.781 pjsua_call.c ..Answering call 1: code=200 .... .... 00:17:01.823 pjsua_media.c ......Audio updated, stream #0: (inactive) .... 00:17:01.834 pjsua_call.c .....Unable to create media session: Unknown pjmedia error 220048 [status=220048] 00:17:01.845 pjsua_core.c ........TX 487 bytes Response msg 488/INVITE/cseq=1652516407
The reason is :
In Callee ( 101) side, audio media is inactive (no match audio codec)
Please add debug log
How to debug:
// in pjsua_call.c, You can print local_sdp
5269 /* Disconnect call after failure in media channel update */
5270 if (status != PJ_SUCCESS) {
#if 1 /* !!!!!!!!!!!!!!! These 3 lines debug print local_sdp */
char buf[4000] = "";
pjmedia_sdp_media_print(local_sdp, buf, sizeof(buf));
PJ_LOG(3, (THIS_FILE, "Local SDP: [%s]\n", buf));
#endif
pjsua_perror(THIS_FILE, "Unable to create media session",
status);
call_disconnect(inv, PJSIP_SC_NOT_ACCEPTABLE_HERE);
/* No need to deinitialize; media will be shutdown when call
* state is disconnected anyway.
*/
/*pjsua_media_channel_deinit(call->index);*/
goto on_return;
}
@jimying I got the information according to the method you gave:
Unable to create media session
00:00:07.344 pjsua_media.c ......Skipped updating media call00:1 (media type=unknown): Unknown pjmedia error 220109
W (8467) pjsua_call.c: call->index = 0
W (8467) sdp.c: media = -���IN
W (8477) sdp.c: port = 32391
W (8477) sdp.c: port_count = -2085978488
W (8477) sdp.c: transport = IN
00:00:07.374 pjsua_call.c .....Local SDP: [m=- 32391/2208988808 IN]
W (8487) sdp.c: media = - W (8497) sdp.c: port = 54290 W (8497) sdp.c: port_count = 1652610066 W (8507) sdp.c: transport = IN 00:00:07.396 pjsua_call.c .....Remote SDP: [m=- 54290/1652610066 IN]
00:00:07.402 pjsua_call.c .....Unable to create media session: Unknown pjmedia error 220048 [status=220048]
And then I don't know what to do with it, you know
@CAIJUN111 Is the debugging code wrong? The printed SDPs (local/remote) are all wrong. I think at least remote-sdp should be right printed
Maybe you can print local & remote sdp before pjsua_media_channel_update(), and try again
// TODO: print local/remote sdp
5256 /* Update media channel with the new SDP */
5257 status = pjsua_media_channel_update(call->index, local_sdp, remote_sdp);
@CAIJUN111 Is the debugging code wrong? The printed SDPs (local/remote) are all wrong. I think at least remote-sdp should be right printed
Maybe you can print local & remote sdp before pjsua_media_channel_update(), and try again
// TODO: print local/remote sdp 5256 /* Update media channel with the new SDP */ 5257 status = pjsua_media_channel_update(call->index, local_sdp, remote_sdp);
@jimying I tried ,the same wrong SDP. The sdps can be found in "INVITE" messge ,I don't think there's anything wrong with it
You need debug print local-sdp to see why no any media codec match. Try another function pjmedia_sdp_neg_get_neg_local() to get local sdp, and print it
In logs: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+Z3/0O95f9G4symYXUmZ5TJ44gZgRAXXQAXHZQFs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:MMhE4PUSf5q7qr6/Soc/D9yeHzcwJ/3ngjm6QvD1
must use SRTP? if use SRTP, need set "use_srtp" and "srtp_secure_signaling" when config account.
In logs: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+Z3/0O95f9G4symYXUmZ5TJ44gZgRAXXQAXHZQFs a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:MMhE4PUSf5q7qr6/Soc/D9yeHzcwJ/3ngjm6QvD1
must use SRTP? if use SRTP, need set "use_srtp" and "srtp_secure_signaling" when config account. @FredSE2021 thanks for your reply
I have not used srtp,you see the message which is PC client's sdp.
I have met "488 Not Acceptable Here" before, when one site use SRTP, but other side not support SRTP. I solved the problem like that: one site to server, use SRTP; server to other side, no SRTP. Maybe PC client need SRTP?
I have met "488 Not Acceptable Here" before, when one site use SRTP, but other side not support SRTP. I solved the problem like that: one site to server, use SRTP; server to other side, no SRTP. Maybe PC client need SRTP? @FredWH But now I am talking to two PJSIP-based devices without SRTP, it still can't communicate
so not SRTP issue.
from logs: [email protected] - > 192.168.1.107 s=pjmedia 192.168.1.107 -> [email protected] s=mss different session name. as jimying said, any more info about media codec?
so not SRTP issue.
from logs: [email protected] - > 192.168.1.107 s=pjmedia 192.168.1.107 -> [email protected] s=mss different session name. as jimying said, any more info about media codec? @FredWH do you kown how to use the function "pjmedia_sdp_media_print()" to print the local sdp? if I can use it correctly,I may kown the reason of wrong
If want to print local sdp, maybe you can try pjmedia_sdp_neg_get_active_local, below files could find pjmedia_sdp_neg_get_active_local usage, pjmedia/src/test/sdp_neg_test.c pjsip-apps/src/samples/simpleua.c
I am sure that 101 is online
蔡俊 @.***
------------------ 原始邮件 ------------------ 发件人: @.>; 发送时间: 2022年5月14日(星期六) 晚上6:22 收件人: @.>; 抄送: @.>; @.>; 主题: Re: [pjsip/pjproject] client with psjip be called by other client,sip server receive "488 Not Acceptable Here" by other sip client. (Issue #3106)
@CAIJUN111
00:00:07.854 pjsua_core.c .RX 531 bytes Response msg 480/INVITE/cseq=21755 (rdata0x3f841c34) from UDP 192.168.1.107:5060: SIP/2.0 480 Temporarily Unavailable
Sip Server response 480, Does callee: 101 already online?
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I don't think this is a PJSIP issue.
Currently the issue page is for bug report or feature request only. For further questions, please use any other means such as StackOverflow.
